Network Working Group S. Floyd
Request for Comments: 2914 ACIRI
BCP: 41 September 2000
Category: Best Current Practice
Congestion Control Principles
Status of this Memo
This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2000). All Rights Reserved.
Abstract
The goal of this document is to explain the need for congestion
control in the Internet, and to discuss what constitutes correct
congestion control. One specific goal is to illustrate the dangers
of neglecting to apply proper congestion control. A second goal is
to discuss the role of the IETF in standardizing new congestion
control protocols.
1. Introduction
This document draws heavily from earlier RFCs, in some cases
reproducing entire sections of the text of earlier documents
[RFC2309, RFC2357]. We have also borrowed heavily from earlier
publications addressing the need for end-to-end congestion control
[FF99].
2. Current standards on congestion control
IETF standards concerning end-to-end congestion control focus either
on specific protocols (e.g., TCP [RFC2581], reliable multicast
protocols [RFC2357]) or on the syntax and semantics of communications
between the end nodes and routers about congestion information (e.g.,
Explicit Congestion Notification [RFC2481]) or desired quality-of-
service (diff-serv)). The role of end-to-end congestion control is
also discussed in an Informational RFC on "Recommendations on Queue
Management and Congestion Avoidance in the Internet" [RFC2309]. RFC
2309 recommends the deployment of active queue management mechanisms
in routers, and the continuation of design efforts towards mechanisms
in routers to deal with flows that are unresponsive to congestion
notification. We freely borrow from RFC 2309 some of their general
discussion of end-to-end congestion control.
In contrast to the RFCs discussed above, this document is a more
general discussion of the principles of congestion control. One of
the keys to the success of the Internet has been the congestion
avoidance mechanisms of TCP. While TCP is still the dominant
transport protocol in the Internet, it is not ubiquitous, and there
are an increasing number of applications that, for one reason or
another, choose not to use TCP. Such traffic includes not only
multicast traffic, but unicast traffic such as streaming multimedia
that does not require reliability; and traffic such as DNS or routing
messages that consist of short transfers deemed critical to the
operation of the network. Much of this traffic does not use any form
of either bandwidth reservations or end-to-end congestion control.
The continued use of end-to-end congestion control by best-effort
traffic is critical for maintaining the stability of the Internet.
This document also discusses the general role of the IETF in the
standardization of new congestion control protocols.
The discussion of congestion control principles for differentiated
services or integrated services is not addressed in this document.
Some categories of integrated or differentiated services include a
guarantee by the network of end-to-end bandwidth, and as such do not
require end-to-end congestion control mechanisms.
3. The development of end-to-end congestion control.
3.1. Preventing congestion collapse.
The Internet protocol architecture is based on a connectionless end-
to-end packet service using the IP protocol. The advantages of its
connectionless design, flexibility and robustness, have been amply
demonstrated. However, these advantages are not without cost:
careful design is required to provide good service under heavy load.
In fact, lack of attention to the dynamics of packet forwarding can
result in severe service degradation or "Internet meltdown". This
phenomenon was first observed during the early growth phase of the
Internet of the mid 1980s [RFC896], and is technically called
"congestion collapse".
The original specification of TCP [RFC793] included window-based flow
control as a means for the receiver to govern the amount of data sent
by the sender. This flow control was used to prevent overflow of the
receiver's data buffer space available for that connection. [RFC793]
reported that segments could be lost due either to errors or to
network congestion, but did not include dynamic adjustment of the
flow-control window in response to congestion.
The original fix for Internet meltdown was provided by Van Jacobson.
Beginning in 1986, Jacobson developed the congestion avoidance
mechanisms that are now required in TCP implementations [Jacobson88,
RFC 2581]. These mechanisms operate in the hosts to cause TCP
connections to "back off" during congestion. We say that TCP flows
are "responsive" to congestion signals (i.e., dropped packets) from
the network. It is these TCP congestion avoidance algorithms that
prevent the congestion collapse of today's Internet.
However, that is not the end of the story. Considerable research has
been done on Internet dynamics since 1988, and the Internet has
grown. It has become clear that the TCP congestion avoidance
mechanisms [RFC2581], while necessary and powerful, are not
sufficient to provide good service in all circumstances. In addition
to the development of new congestion control mechanisms [RFC2357],
router-based mechanisms are in development that complement the
endpoint congestion avoidance mechanisms.
A major issue that still needs to be addressed is the potential for
future congestion collapse of the Internet due to flows that do not
use responsible end-to-end congestion control. RFC 896 [RFC896]
suggested in 1984 that gateways should detect and `squelch'
misbehaving hosts: "Failure to respond to an ICMP Source Quench
message, though, should be regarded as grounds for action by a
gateway to disconnect a host. Detecting such failure is non-trivial
but is a worthwhile area for further research." Current papers
still propose that routers detect and penalize flows that are not
employing acceptable end-to-end congestion control [FF99].
3.2. Fairness
In addition to a concern about congestion collapse, there is a
concern about `fairness' for best-effort traffic. Because TCP "backs
off" during congestion, a large number of TCP connections can share a
single, congested link in such a way that bandwidth is shared
reasonably equitably among similarly situated flows. The equitable
sharing of bandwidth among flows depends on the fact that all flows
are running compatible congestion control algorithms. For TCP, this
means congestion control algorithms conformant with the current TCP
specification [RFC793, RFC1122, RFC2581].
The issue of fairness among competing flows has become increasingly
important for several reasons. First, using window scaling
[RFC1323], individual TCPs can use high bandwidth even over high-
propagation-delay paths. Second, with the growth of the web,
Internet users increasingly want high-bandwidth and low-delay
communications, rather than the leisurely transfer of a long file in
the background. The growth of best-effort traffic that does not use
TCP underscores this concern about fairness between competing best-
effort traffic in times of congestion.
The popularity of the Internet has caused a proliferation in the
number of TCP implementations. Some of these may fail to implement
the TCP congestion avoidance mechanisms correctly because of poor
implementation [RFC2525]. Others may deliberately be implemented
with congestion avoidance algorithms that are more aggressive in
their use of bandwidth than other TCP implementations; this would
allow a vendor to claim to have a "faster TCP". The logical
consequence of such implementations would be a spiral of increasingly
aggressive TCP implementations, or increasingly aggressive transport
protocols, leading back to the point where there is effectively no
congestion avoidance and the Internet is chronically congested.
There is a well-known way to achieve more aggressive performance
without even changing the transport protocol, by changing the level
of granularity: open multiple connections to the same place, as has
been done in the past by some Web browsers. Thus, instead of a
spiral of increasingly aggressive transport protocols, we would
instead have a spiral of increasingly aggressive web browsers, or
increasingly aggressive applications.
This raises the issue of the appropriate granularity of a "flow",
where we define a `flow' as the level of granularity appropriate for
the application of both fairness and congestion control. From RFC
2309: "There are a few `natural' answers: 1) a TCP or UDP connection
(source address/port, destination address/port); 2) a
source/destination host pair; 3) a given source host or a given
destination host. We would guess that the source/destination host
pair gives the most appropriate granularity in many circumstances.
The granularity of flows for congestion management is, at least in
part, a policy question that needs to be addressed in the wider IETF
community."
Again borrowing from RFC 2309, we use the term "TCP-compatible" for a
flow that behaves under congestion like a flow produced by a
conformant TCP. A TCP-compatible flow is responsive to congestion
notification, and in steady-state uses no more bandwidth than a
conformant TCP running under comparable conditions (drop rate, RTT,
MTU, etc.)
It is convenient to divide flows into three classes: (1) TCP-
compatible flows, (2) unresponsive flows, i.e., flows that do not
slow down when congestion occurs, and (3) flows that are responsive
but are not TCP-compatible. The last two classes contain more
aggressive flows that pose significant threats to Internet
performance, as we discuss below.
In addition to steady-state fairness, the fairness of the initial
slow-start is also a concern. One concern is the transient effect on
other flows of a flow with an overly-aggressive slow-start procedure.
Slow-start performance is particularly important for the many flows
that are short-lived, and only have a small amount of data to
transfer.
3.3. Optimizing performance regarding throughput, delay, and loss.
In addition to the prevention of congestion collapse and concerns
about fairness, a third reason for a flow to use end-to-end
congestion control can be to optimize its own performance regarding
throughput, delay, and loss. In some circumstances, for example in
environments of high statistical multiplexing, the delay and loss
rate experienced by a flow are largely independent of its own sending
rate. However, in environments with lower levels of statistical
multiplexing or with per-flow scheduling, the delay and loss rate
experienced by a flow is in part a function of the flow's own sending
rate. Thus, a flow can use end-to-end congestion control to limit
the delay or loss experienced by its own packets. We would note,
however, that in an environment like the current best-effort
Internet, concerns regarding congestion collapse and fairness with
competing flows limit the range of congestion control behaviors
available to a flow.
4. The role of the standards process
The standardization of a transport protocol includes not only
standardization of aspects of the protocol that could affect
interoperability (e.g., information exchanged by the end-nodes), but
also standardization of mechanisms deemed critical to performance
(e.g., in TCP, reduction of the congestion window in response to a
packet drop). At the same time, implementation-specific details and
other aspects of the transport protocol that do not affect
interoperability and do not significantly interfere with performance
do not require standardization. Areas of TCP that do not require
standardization include the details of TCP's Fast Recovery procedure
after a Fast Retransmit [RFC2582]. The appendix uses examples from
TCP to discuss in more detail the role of the standards process in
the development of congestion control.
4.1. The development of new transport protocols.
In addition to addressing the danger of congestion collapse, the
standardization process for new transport protocols takes care to
avoid a congestion control `arms race' among competing protocols. As
an example, in RFC 2357 [RFC2357] the TSV Area Directors and their
Directorate outline criteria for the publication as RFCs of
Internet-Drafts on reliable multicast transport protocols. From
[RFC2357]: "A particular concern for the IETF is the impact of
reliable multicast traffic on other traffic in the Internet in times
of congestion, in particular the effect of reliable multicast traffic
on competing TCP traffic.... The challenge to the IETF is to
encourage research and implementations of reliable multicast, and to
enable the needs of applications for reliable multicast to be met as
expeditiously as possible, while at the same time protecting the
Internet from the congestion disaster or collapse that could result
from the widespread use of applications with inappropriate reliable
multicast mechanisms."
The list of technical criteria that must be addressed by RFCs on new
reliable multicast transport protocols include the following: "Is
there a congestion control mechanism? How well does it perform? When
does it fail? Note that congestion control mechanisms that operate
on the network more aggressively than TCP will face a great burden of
proof that they don't threaten network stability."
It is reasonable to expect that these concerns about the effect of
new transport protocols on competing traffic will apply not only to
reliable multicast protocols, but to unreliable unicast, reliable
unicast, and unreliable multicast traffic as well.
4.2. Application-level issues that affect congestion control
The specific issue of a browser opening multiple connections to the
same destination has been addressed by RFC 2616 [RFC2616], which
states in Section 8.1.4 that "Clients that use persistent connections
SHOULD limit the number of simultaneous connections that they
maintain to a given server. A single-user client SHOULD NOT maintain
more than 2 connections with any server or proxy."
4.3. New developments in the standards process
The most obvious developments in the IETF that could affect the
evolution of congestion control are the development of integrated and
differentiated services [RFC2212, RFC2475] and of Explicit Congestion
Notification (ECN) [RFC2481]. However, other less dramatic
developments are likely to affect congestion control as well.
One such effort is that to construct Endpoint Congestion Management
[BS00], to enable multiple concurrent flows from a sender to the same
receiver to share congestion control state. By allowing multiple
connections to the same destination to act as one flow in terms of
end-to-end congestion control, a Congestion Manager could allow
individual connections slow-starting to take advantage of previous
information about the congestion state of the end-to-end path.
Further, the use of a Congestion Manager could remove the congestion
control dangers of multiple flows being opened between the same
source/destination pair, and could perhaps be used to allow a browser
to open many simultaneous connections to the same destination.
5. A description of congestion collapse
This section discusses congestion collapse from undelivered packets
in some detail, and shows how unresponsive flows could contribute to
congestion collapse in the Internet. This section draws heavily on
material from [FF99].
Informally, congestion collapse occurs when an increase in the
network load results in a decrease in the useful work done by the
network. As discussed in Section 3, congestion collapse was first
reported in the mid 1980s [RFC896], and was largely due to TCP
connections unnecessarily retransmitting packets that were either in
transit or had already been received at the receiver. We call the
congestion collapse that results from the unnecessary retransmission
of packets classical congestion collapse. Classical congestion
collapse is a stable condition that can result in throughput that is
a small fraction of normal [RFC896]. Problems with classical
congestion collapse have generally been corrected by the timer
improvements and congestion control mechanisms in modern
implementations of TCP [Jacobson88].
A second form of potential congestion collapse occurs due to
undelivered packets. Congestion collapse from undelivered packets
arises when bandwidth is wasted by delivering packets through the
network that are dropped before reaching their ultimate destination.
This is probably the largest unresolved danger with respect to
congestion collapse in the Internet today. Different scenarios can
result in different degrees of congestion collapse, in terms of the
fraction of the congested links' bandwidth used for productive work.
The danger of congestion collapse from undelivered packets is due
primarily to the increasing deployment of open-loop applications not
using end-to-end congestion control. Even more destructive would be
best-effort applications that *increase* their sending rate in
response to an increased packet drop rate (e.g., automatically using
an increased level of FEC).
Table 1 gives the results from a scenario with congestion collapse
from undelivered packets, where scarce bandwidth is wasted by packets
that never reach their destination. The simulation uses a scenario
with three TCP flows and one UDP flow competing over a congested 1.5
Mbps link. The access links for all nodes are 10 Mbps, except that
the access link to the receiver of the UDP flow is 128 Kbps, only 9%
of the bandwidth of shared link. When the UDP source rate exceeds
128 Kbps, most of the UDP packets will be dropped at the output port
to that final link.
UDP
Arrival UDP TCP Total
Rate Goodput Goodput Goodput
--------------------------------------
0.7 0.7 98.5 99.2
1.8 1.7 97.3 99.1
2.6 2.6 96.0 98.6
5.3 5.2 92.7 97.9
8.8 8.4 87.1 95.5
10.5 8.4 84.8 93.2
13.1 8.4 81.4 89.8
17.5 8.4 77.3 85.7
26.3 8.4 64.5 72.8
52.6 8.4 38.1 46.4
58.4 8.4 32.8 41.2
65.7 8.4 28.5 36.8
75.1 8.4 19.7 28.1
87.6 8.4 11.3 19.7
105.2 8.4 3.4 11.8
131.5 8.4 2.4 10.7
Table 1. A simulation with three TCP flows and one UDP flow.
Table 1 shows the UDP arrival rate from the sender, the UDP goodput
(defined as the bandwidth delivered to the receiver), the TCP goodput
(as delivered to the TCP receivers), and the aggregate goodput on the
congested 1.5 Mbps link. Each rate is given as a fraction of the
bandwidth of the congested link. As the UDP source rate increases,
the TCP goodput decreases roughly linearly, and the UDP goodput is
nearly constant. Thus, as the UDP flow increases its offered load,
its only effect is to hurt the TCP and aggregate goodput. On the
congested link, the UDP flow ultimately `wastes' the bandwidth that
could have been used by the TCP flow, and reduces the goodput in the
network as a whole down to a small fraction of the bandwidth of the
congested link.
The simulations in Table 1 illustrate both unfairness and congestion
collapse. As [FF99] discusses, compatible congestion control is not
the only way to provide fairness; per-flow scheduling at the
congested routers is an alternative mechanism at the routers that
guarantees fairness. However, as discussed in [FF99], per-flow
scheduling can not be relied upon to prevent congestion collapse.
There are only two alternatives for eliminating the danger of
congestion collapse from undelivered packets. The first alternative
for preventing congestion collapse from undelivered packets is the
use of effective end-to-end congestion control by the end nodes.
More specifically, the requirement would be that a flow avoid a
pattern of significant losses at links downstream from the first
congested link on the path. (Here, we would consider any link a
`congested link' if any flow is using bandwidth that would otherwise
be used by other traffic on the link.) Given that an end-node is
generally unable to distinguish between a path with one congested
link and a path with multiple congested links, the most reliable way
for a flow to avoid a pattern of significant losses at a downstream
congested link is for the flow to use end-to-end congestion control,
and reduce its sending rate in the presence of loss.
A second alternative for preventing congestion collapse from
undelivered packets would be a guarantee by the network that packets
accepted at a congested link in the network will be delivered all the
way to the receiver [RFC2212, RFC2475]. We note that the choice
between the first alternative of end-to-end congestion control and
the second alternative of end-to-end bandwidth guarantees does not
have to be an either/or decision; congestion collapse can be
prevented by the use of effective end-to-end congestion by some of
the traffic, and the use of end-to-end bandwidth guarantees from the
network for the rest of the traffic.
6. Forms of end-to-end congestion control
This document has discussed concerns about congestion collapse and
about fairness with TCP for new forms of congestion control. This
does not mean, however, that concerns about congestion collapse and
fairness with TCP necessitate that all best-effort traffic deploy
congestion control based on TCP's Additive-Increase Multiplicative-
Decrease (AIMD) algorithm of reducing the sending rate in half in
response to each packet drop. This section separately discusses the
implications of these two concerns of congestion collapse and
fairness with TCP.
6.1. End-to-end congestion control for avoiding congestion collapse.
The avoidance of congestion collapse from undelivered packets
requires that flows avoid a scenario of a high sending rate, multiple
congested links, and a persistent high packet drop rate at the
downstream link. Because congestion collapse from undelivered
packets consists of packets that waste valuable bandwidth only to be
dropped downstream, this form of congestion collapse is not possible
in an environment where each flow traverses only one congested link,
or where only a small number of packets are dropped at links
downstream of the first congested link. Thus, any form of congestion
control that successfully avoids a high sending rate in the presence
of a high packet drop rate should be sufficient to avoid congestion
collapse from undelivered packets.
We would note that the addition of Explicit Congestion Notification
(ECN) to the IP architecture would not, in and of itself, remove the
danger of congestion collapse for best-effort traffic. ECN allows
routers to set a bit in packet headers as an indication of congestion
to the end-nodes, rather than being forced to rely on packet drops to
indicate congestion. However, with ECN, packet-marking would replace
packet-dropping only in times of moderate congestion. In particular,
when congestion is heavy, and a router's buffers overflow, the router
has no choice but to drop arriving packets.
6.2. End-to-end congestion control for fairness with TCP.
The concern expressed in [RFC2357] about fairness with TCP places a
significant though not crippling constraint on the range of viable
end-to-end congestion control mechanisms for best-effort traffic. An
environment with per-flow scheduling at all congested links would
isolate flows from each other, and eliminate the need for congestion
control mechanisms to be TCP-compatible. An environment with
differentiated services, where flows marked as belonging to a certain
diff-serv class would be scheduled in isolation from best-effort
traffic, could allow the emergence of an entire diff-serv class of
traffic where congestion control was not required to be TCP-
compatible. Similarly, a pricing-controlled environment, or a diff-
serv class with its own pricing paradigm, could supercede the concern
about fairness with TCP. However, for the current Internet
environment, where other best-effort traffic could compete in a FIFO
queue with TCP traffic, the absence of fairness with TCP could lead
to one flow `starving out' another flow in a time of high congestion,
as was illustrated in Table 1 above.
However, the list of TCP-compatible congestion control procedures is
not limited to AIMD with the same increase/ decrease parameters as
TCP. Other TCP-compatible congestion control procedures include
rate-based variants of AIMD; AIMD with different sets of
increase/decrease parameters that give the same steady-state
behavior; equation-based congestion control where the sender adjusts
its sending rate in response to information about the long-term
packet drop rate; layered multicast where receivers subscribe and
unsubscribe from layered multicast groups; and possibly other forms
that we have not yet begun to consider.
7. Acknowledgements
Much of this document draws directly on previous RFCs addressing
end-to-end congestion control. This attempts to be a summary of
ideas that have been discussed for many years, and by many people.
In particular, acknowledgement is due to the members of the End-to-
End Research Group, the Reliable Multicast Research Group, and the
Transport Area Directorate. This document has also benefited from
discussion and feedback from the Transport Area Working Group.
Particular thanks are due to Mark Allman for feedback on an earlier
version of this document.
8. References
[BS00] Balakrishnan H. and S. Seshan, "The Congestion Manager",
Work in Progress.
[DMKM00] Dawkins, S., Montenegro, G., Kojo, M. and V. Magret,
"End-to-end Performance Implications of Slow Links",
Work in Progress.
[FF99] Floyd, S. and K. Fall, "Promoting the Use of End-to-End
Congestion Control in the Internet", IEEE/ACM
Transactions on Networking, August 1999. URL
http://www.aciri.org/floyd/end2end-paper.html
[HPF00] Handley, M., Padhye, J. and S. Floyd, "TCP Congestion
Window Validation", RFC 2861, June 2000.
[Jacobson88] V. Jacobson, Congestion Avoidance and Control, ACM
SIGCOMM '88, August 1988.
[RFC793] Postel, J., "Transmission Control Protocol", STD 7, RFC
793, September 1981.
[RFC896] Nagle, J., "Congestion Control in IP/TCP", RFC 896,
January 1984.
[RFC1122] Braden, R., Ed., "Requirements for Internet Hosts --
Communication Layers", STD 3, RFC 1122, October 1989.
[RFC1323] Jacobson, V., Braden, R. and D. Borman, "TCP Extensions
for High Performance", RFC 1323, May 1992.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2212] Shenker, S., Partridge, C. and R. Guerin, "Specification
of Guaranteed Quality of Service", RFC 2212, September
1997.
[RFC2309] Braden, R., Clark, D., Crowcroft, J., Davie, B.,
Deering, S., Estrin, D., Floyd, S., Jacobson, V.,
Minshall, G., Partridge, C., Peterson, L., Ramakrishnan,
K.K., Shenker, S., Wroclawski, J., and L. Zhang,
"Recommendations on Queue Management and Congestion
Avoidance in the Internet", RFC 2309, April 1998.
[RFC2357] Mankin, A., Romanow, A., Bradner, S. and V. Paxson,
"IETF Criteria for Evaluating Reliable Multicast
Transport and Application Protocols", RFC 2357, June
1998.
[RFC2414] Allman, M., Floyd, S. and C. Partridge, "Increasing
TCP's Initial Window", RFC 2414, September 1998.
[RFC2475] Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.
and W. Weiss, "An Architecture for Differentiated
Services", RFC 2475, December 1998.
[RFC2481] Ramakrishnan K. and S. Floyd, "A Proposal to add
Explicit Congestion Notification (ECN) to IP", RFC 2481,
January 1999.
[RFC2525] Paxson, V., Allman, M., Dawson, S., Fenner, W., Griner,
J., Heavens, I., Lahey, K., Semke, J. and B. Volz,
"Known TCP Implementation Problems", RFC 2525, March
1999.
[RFC2581] Allman, M., Paxson, V. and W. Stevens, "TCP Congestion
Control", RFC 2581, April 1999.
[RFC2582] Floyd, S. and T. Henderson, "The NewReno Modification to
TCP's Fast Recovery Algorithm", RFC 2582, April 1999.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P. and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
[SCWA99] S. Savage, N. Cardwell, D. Wetherall, and T. Anderson,
TCP Congestion Control with a Misbehaving Receiver, ACM
Computer Communications Review, October 1999.
[TCPB98] Hari Balakrishnan, Venkata N. Padmanabhan, Srinivasan
Seshan, Mark Stemm, and Randy H. Katz, TCP Behavior of a
Busy Internet Server: Analysis and Improvements, IEEE
Infocom, March 1998. Available from:
"http://www.cs.berkeley.edu/~hari/papers/infocom98.ps.gz".
[TCPF98] Dong Lin and H.T. Kung, TCP Fast Recovery Strategies:
Analysis and Improvements, IEEE Infocom, March 1998.
Available from:
"http://www.eecs.harvard.edu/networking/papers/infocom-
tcp-final-198.pdf".
9. TCP-Specific issues
In this section we discuss some of the particulars of TCP congestion
control, to illustrate a realization of the congestion control
principles, including some of the details that arise when
incorporating them into a production transport protocol.
9.1. Slow-start.
The TCP sender can not open a new connection by sending a large burst
of data (e.g., a receiver's advertised window) all at once. The TCP
sender is limited by a small initial value for the congestion window.
During slow-start, the TCP sender can increase its sending rate by at
most a factor of two in one roundtrip time. Slow-start ends when
congestion is detected, or when the sender's congestion window is
greater than the slow-start threshold ssthresh.
An issue that potentially affects global congestion control, and
therefore has been explicitly addressed in the standards process,
includes an increase in the value of the initial window
[RFC2414,RFC2581].
Issues that have not been addressed in the standards process, and are
generally considered not to require standardization, include such
issues as the use (or non-use) of rate-based pacing, and mechanisms
for ending slow-start early, before the congestion window reaches
ssthresh. Such mechanisms result in slow-start behavior that is as
conservative or more conservative than standard TCP.
9.2. Additive Increase, Multiplicative Decrease.
In the absence of congestion, the TCP sender increases its congestion
window by at most one packet per roundtrip time. In response to a
congestion indication, the TCP sender decreases its congestion window
by half. (More precisely, the new congestion window is half of the
minimum of the congestion window and the receiver's advertised
window.)
An issue that potentially affects global congestion control, and
therefore would be likely to be explicitly addressed in the standards
process, would include a proposed addition of congestion control for
the return stream of `pure acks'.
An issue that has not been addressed in the standards process, and is
generally not considered to require standardization, would be a
change to the congestion window to apply as an upper bound on the
number of bytes presumed to be in the pipe, instead of applying as a
sliding window starting from the cumulative acknowledgement.
(Clearly, the receiver's advertised window applies as a sliding
window starting from the cumulative acknowledgement field, because
packets received above the cumulative acknowledgement field are held
in TCP's receive buffer, and have not been delivered to the
application. However, the congestion window applies to the number of
packets outstanding in the pipe, and does not necessarily have to
include packets that have been received out-of-order by the TCP
receiver.)
9.3. Retransmit timers.
The TCP sender sets a retransmit timer to infer that a packet has
been dropped in the network. When the retransmit timer expires, the
sender infers that a packet has been lost, sets ssthresh to half of
the current window, and goes into slow-start, retransmitting the lost
packet. If the retransmit timer expires because no acknowledgement
has been received for a retransmitted packet, the retransmit timer is
also "backed-off", doubling the value of the next retransmit timeout
interval.
An issue that potentially affects global congestion control, and
therefore would be likely to be explicitly addressed in the standards
process, might include a modified mechanism for setting the
retransmit timer that could significantly increase the number of
retransmit timers that expire prematurely, when the acknowledgement
has not yet arrived at the sender, but in fact no packets have been
dropped. This could be of concern to the Internet standards process
because retransmit timers that expire prematurely could lead to an
increase in the number of packets unnecessarily transmitted on a
congested link.
9.4. Fast Retransmit and Fast Recovery.
After seeing three duplicate acknowledgements, the TCP sender infers
a packet loss. The TCP sender sets ssthresh to half of the current
window, reduces the congestion window to at most half of the previous
window, and retransmits the lost packet.
An issue that potentially affects global congestion control, and
therefore would be likely to be explicitly addressed in the standards
process, might include a proposal (if there was one) for inferring a
lost packet after only one or two duplicate acknowledgements. If
poorly designed, such a proposal could lead to an increase in the
number of packets unnecessarily transmitted on a congested path.
An issue that has not been addressed in the standards process, and
would not be expected to require standardization, would be a proposal
to send a "new" or presumed-lost packet in response to a duplicate or
partial acknowledgement, if allowed by the congestion window. An
example of this would be sending a new packet in response to a single
duplicate acknowledgement, to keep the `ack clock' going in case no
further acknowledgements would have arrived. Such a proposal is an
example of a beneficial change that does not involve interoperability
and does not affect global congestion control, and that therefore
could be implemented by vendors without requiring the intervention of
the IETF standards process. (This issue has in fact been addressed
in [DMKM00], which suggests that "researchers may wish to experiment
with injecting new traffic into the network when duplicate
acknowledgements are being received, as described in [TCPB98] and
[TCPF98]."
9.5. Other aspects of TCP congestion control.
Other aspects of TCP congestion control that have not been discussed
in any of the sections above include TCP's recovery from an idle or
application-limited period [HPF00].
10. Security Considerations
This document has been about the risks associated with congestion
control, or with the absence of congestion control. Section 3.2
discusses the potentials for unfairness if competing flows don't use
compatible congestion control mechanisms, and Section 5 considers the
dangers of congestion collapse if flows don't use end-to-end
congestion control.
Because this document does not propose any specific congestion
control mechanisms, it is also not necessary to present specific
security measures associated with congestion control. However, we
would note that there are a range of security considerations
associated with congestion control that should be considered in IETF
documents.
For example, individual congestion control mechanisms should be as
robust as possible to the attempts of individual end-nodes to subvert
end-to-end congestion control [SCWA99]. This is a particular concern
in multicast congestion control, because of the far-reaching
distribution of the traffic and the greater opportunities for
individual receivers to fail to report congestion.
RFC 2309 also discussed the potential dangers to the Internet of
unresponsive flows, that is, flows that don't reduce their sending
rate in the presence of congestion, and describes the need for
mechanisms in the network to deal with flows that are unresponsive to
congestion notification. We would note that there is still a need
for research, engineering, measurement, and deployment in these
areas.
Because the Internet aggregates very large numbers of flows, the risk
to the whole infrastructure of subverting the congestion control of a
few individual flows is limited. Rather, the risk to the
infrastructure would come from the widespread deployment of many
end-nodes subverting end-to-end congestion control.
AUTHOR'S ADDRESS
Sally Floyd
AT&T Center for Internet Research at ICSI (ACIRI)
Phone: +1 (510) 642-4274 x189
EMail: floyd@aciri.org
URL: http://www.aciri.org/floyd/
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